Rockbox only uses the first album art image (APIC / PIC frame) found in id3v2
tags. When a file contains more than one image the second one is ignored but
the parsealbumart() callback overwrites the already set data. This causes the
metadata structure to contain an invalid pointer to the image data, resulting
in no image shown.
Make parsealbumart() aware of this and skip parsing when an albumart image has
already been found. Fixes FS#12870.
Change-Id: Id8164f319cd5e1ee868b581f8f4ad3ea69c17f77
Most SoCs are these days are fast enough for realtime BRR, gaussian
interpolation and echo processing.
Change-Id: I180ce8ad45242c67b5e573a406b9522098a3f12b
Affected BRR cached waveforms but not realtime BRR decode as far as
I could ascertain. BRR cached waves required loop points to be inside
the initial waveform but this change removes that restriction.
Change-Id: I0ef4db720e5c28bd7b2fb9ae255d27c0a7213f79
CPU optimization gets its own files in which to fill-in optimizable
routines.
Some pointless #if 0's for profiling need removal. Those macros are
empty if not profiling.
Force some functions that are undesirable to be force-inlined by the
compiler to be not inlined.
Change-Id: Ia7b7e45380d7efb20c9b1a4d52e05db3ef6bbaab
Use the tlsf malloc and friends instead of the silly
codec_malloc to get actually working free and saner
realloc that doesn't leak memory.
Makes files with moderately sized embedded AA play
on targets with large enough codec buffers and files
with too large AA are now skipped rather than crashing.
Fixes crash when playing example file in FS#12842.
Change-Id: I06562955c4d9a95bd90f55738214fba462092b71
Uses the Catmull-Rom case of Hermite cubic splines.
Vastly improves the quality and accuracy of audio resampling with a
rather minor additional overhead compared to the previous linear
implementation.
ARM and Coldfire assembly implementations included.
Change-Id: Ic45d84bc66c5b312ef373198297a952167a4be26
Reviewed-on: http://gerrit.rockbox.org/304
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
DSP_SWITCH_FREQUENCY has been deprecated and the same enumerated value
as DSP_SET_FREQUENCY since major DSP revisions were committed. This
task should have been performed much earlier but, oh well, do it now.
Change-Id: I3f30d651b894136a07c7e17f78fc16a7d98631ff
Input type can only change once per call because the DSP parameters
are only copied at the start and input is always taken from the src
buffer which means sample input format switching can be once per call
instead of once per loop.
Change-Id: Ifa3521753428fb0e6997e4934f24a3b915628cc7
Some things can just be a bit simpler in handling the list of stages
and some things, especially format change handling, can be simplified
for each stage implementation. Format changes are sent through the
configure() callback.
Hide some internal details and variables from processing stages and
let the core deal with it.
Do some miscellaneous cleanup and keep things a bit better factored.
Change-Id: I19dd8ce1d0b792ba914d426013088a49a52ecb7e
When seeking to the next id3v2 frame we need to consider if the tag has the
unsync flag set. Not doing so will likely make parsing end up in the middle of
the current frame if the frame size exceeds the upper limit set during read.
The latter usually happens for album art frames.
Fixes FS#12849.
Change-Id: Ic92853eef4374508d84df347bcc66b6661d5037d
This is going right in since it's long overdue. If anything is goofed,
drop me a line or just tweak it yourself if you know what's wrong. :-)
Make HW/SW codec interface more uniform when emulating HW functionality
on SWCODEC for functions such as "audiohw_set_pitch". The firmware-to-
DSP plumbing is in firmware/drivers/audiohw-swcodec.c. "sound_XXX"
APIs are all in sound.c with none in DSP code any longer.
Reduce number of settings definitions needed by each codec by providing
defaults for common ones like balance, channels and SW tone controls.
Remove need for separate SIM code and tables and add virtual codec header
for hosted targets.
Change-Id: I3f23702bca054fc9bda40f49824ce681bb7f777b
Implements double-buffered volume, balance and prescaling control in
the main PCM driver when HAVE_SW_VOLUME_CONTROL is defined ensuring
that all PCM is volume controlled and level changes are low in latency.
Supports -73 to +6 dB using a 15-bit factor so that no large-integer
math is needed.
Low-level hardware drivers do not have to implement it themselves but
parameters can be changed (currently defined in pcm-internal.h) to work
best with a particular SoC or to provide different volume ranges.
Volume and prescale calls should be made in the codec driver. It should
appear as a normal hardware interface. PCM volume calls expect .1 dB
units.
Change-Id: Idf6316a64ef4fb8abcede10707e1e6c6d01d57db
Reviewed-on: http://gerrit.rockbox.org/423
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
Will need it soon enough.
Combine the contents of all the various fixedpoint.h files.
Not moving fixedpoint.c for now since I'm not sure where it
should be and it causes some dependency issues.
Change-Id: Ideacbca2ca78f9158c2b114b113c274f68e908d5
Prevents cutoff of tracks, especially short ones:
* Extend looped tracks by fade length to fade at start of loop repeat.
* No fade occurs for non-repeating track only having an intro.
* Uses id3.tail_trim field to store fade duration.
Use libGME built-in elapsed time reporting instead of custom calculation:
* libGME already reports in milliseconds.
* Don't advance time counter when Repeat == One. It just runs the progress
over the length limit.
Fix a comment about sample rate and set the reported bitrate to be
accurate for 44.1 kHz stereo.
Change-Id: I3ede22bda0f9a941a3fef751f4d678eb0027344c
Flush decoder state and frame out buffer upon a forced stop to prevent
a short burst of stale audio from the previously decoding track from
playing when skipping from one WMA track to another.
Change-Id: I24c910c5dbd83caed2510db68d9e39a474332a79
Reviewed-on: http://gerrit.rockbox.org/406
Reviewed-by: Michael Sevakis <jethead71@rockbox.org>
Tested-by: Michael Sevakis <jethead71@rockbox.org>
Instead of 3 cfg lines per eq band there is now a single line
for each:
<config name>: <cutoff/center freq>, <q>, <gain>
In addition, the config value names make a bit more sense.
The old settings are still readable but config.cfg and any new
settings files will be written with the new config values. (The
old settings will be removed completly sometime after the next
stable release).
Also a slight rework of the advanced EQ menu UI
Change-Id: I9008658d36ded442a5f2f825916df42a3934cbef
Reviewed-on: http://gerrit.rockbox.org/394
Reviewed-by: Jonathan Gordon <rockbox@jdgordon.info>
- A 10 Band EQ for Rockbox w/ presets adapted
from VLC
- frequency stepping at 32, 64, 125, 250, 500
1K, 2K, 4K, 8K, 16K
Change-Id: I85ad84d70a534edfc66c6ad9af8a76f022a02ec7
Reviewed-on: http://gerrit.rockbox.org/386
Reviewed-by: Jonathan Gordon <rockbox@jdgordon.info>
Speeds up decoding of 128k opus files by 1.2MHz on AMSv2. Rounding
error is 1 bit due to KissFFT using a 15 bit shift instead of a 16 bit shift.
Also, change an LDMIA in the armv4 code to LDM as the pointer should not
increment.
Change-Id: I626a207c6a056a1984e33cfe89415c35d0caed93
Reviewed-on: http://gerrit.rockbox.org/377
Reviewed-by: Michael Giacomelli <giac2000@hotmail.com>
Tested-by: Michael Giacomelli <giac2000@hotmail.com>
The old GCC version currently required (sbox-arm-linux-gcc 3.4.4
release) apparently has trouble with function pointers used as
static array initializers when using indexed initializers + ranges
(ie. [A ... B] = fn).
Change-Id: I494c2b607e4d93a9893264749d0ac257fb54ce3b
I'm not 100% sure that the rounding of denormals is correct. As compared to foobar2000,
some samples are off by +1 LSB. However, since I can't output 24 bit PCM easily with
rockbox, I'm not sure if this is due to a bug or just how rockbox rounds. In practice
I don't think it matters so I'm just going to commit this for now.
Change-Id: Ic0792fcb172e4369a5512d202121c2b918b36079
avoids complicated index calculations in the loops.
saves 0.3MHz decoding a 64kbps test file on h300 (cf) and
0.2MHz on c200 (pp)
Change-Id: I1918912d9a4502f89980c6bb270ec2ef10a07010
Signed-off-by: Nils Wallménius <nils@rockbox.org>
on a target with a disk.
Change-Id: I37c875c9cd014eb61fe5232dab0f4b8f15f057dd
Reviewed-on: http://gerrit.rockbox.org/319
Tested-by: Thiago Okada <thiago.mast3r@gmail.com>
Reviewed-by: Frederik Vestre <freqmod@gmail.com>
Tested-by: Frederik Vestre <freqmod@gmail.com>
speeds up decoding of a 64kbps test_file by 1.5MHz on c200 (pp)
and 1.9MHz on fuzev1 (amsv1)
Change-Id: I1db460b634eba608c3e00541d96fc93d5a05710b
Signed-off-by: Nils Wallménius <nils@rockbox.org>
speeds up decoding of a 64kbps test file by 0.5Hz on h300 (cf)
0.9MHz on c200 (pp) and 0.2MHz on fuzev1 (amsv1)
Change-Id: Ib537c2393fa6dca0b61e4e9f80eef5e688c2c2bd
Signed-off-by: Nils Wallménius <nils@rockbox.org>
Unroll overlap add loop by four and use memcpy for copying
instead of loops.
Change-Id: I17114626a395d5972130251d892f851bc86e3a6a
Signed-off-by: Nils Wallménius <nils@rockbox.org>
Replace complicated macro doing three 16*16 muls and add an inline
asm implementation for arm, speeds up decoding a 64kbps test file
by 0.5MHz on c200 (pp) and gives slightly better precision.
Change-Id: I6fc5b83c210f01bffdc38aec54cc5a8b646d8169
Signed-off-by: Nils Wallménius <nils@rockbox.org>
Hoist load of coefficients out of the loop.
Speeds up decoding of a 64kbps test file by 0.6MHz on h300 (cf)
0.2MHz on c200 (pp) and 0.1MHz on fuzev1 (amsv1)
Signed-off-by: Nils Wallménius <nils@rockbox.org>
Change-Id: I4be0059fc2a77748575f5fc9378f7f348d64f1c4
